In traditional circuit-switched networks, such as the Public Switched Telephony Network (“PSTN”), each user endpoint is connected to at most one switching system. In a business enterprise, a business telephone is connected to a single Private Branch Exchange (“PBX”). A PBX is an intelligent switching point within a circuit-switched network that is responsible for routing calls to a plurality of internal nodes or public destinations via a single PSTN switching system.
Newer telephony networks that employ packet-switching technologies are growing in popularity. In particular, packet-switched telephony networks that use the Internet Protocol (IP) as a network protocol for transmitting and receiving voice data are becoming more prevalent. These so-called Internet Telephony (IT) networks have the potential to offer new features and services that are currently unavailable to subscribers of circuit-switched telephony networks. Conceptually, IT Networks differ from the PSTN systems in that they generally transmit voice data exchanged between two subscriber endpoints, according to an IP format. More specifically, they encapsulate voice data into data packets, which are transmitted according to an IP format in a similar manner as textual data is transmitted from one computer to another via the internet.
The Session Initiation Protocol (SIP) is one of several protocols that may be used with the Internet Protocol to support Internet Telephony applications. The SIP specification is defined in the Internet Engineering Task Force (IETF) Request for Comments (RFC) 3261, dated June 2002; the disclosure of which is incorporated herein by reference in its entirety. SIP is an application-layer control protocol for creating, modifying, and terminating sessions between networked endpoints, which are referred to as SIP Enabled Devices, User Agents or simply SIP endpoints.
As discussed above, SIP Enabled Devices implement a network communication protocol, wherein a communication session is established for two endpoints to transmit and receive data. As such, each SIP Device in a SIP network is assigned a unique SIP address or terminal name, which is defined in a SIP Universal Resource Identifier (URI). The format of a SIP URI is similar to that of an email address, which typically includes a user name “@” a domain name, for example “sip:alice@siemens.com.” SIP URI data is placed into header fields of SIP messages, for example to identify a sender and a receiver of the SIP message. For secure communications, the SIP Specification also defines a SIPS URI, for example “sips:alice@siemens.com.” Accordingly, when a SIPS URI is used the SIP Enabled Device associated with the SIP URI may implement an encryption protocol for transmitting data in a secure communication session. It should be noted that the SIPS URI protocol may be used in the same way as the SIP URI
The mechanism to establish secure voice over IP communication calls involves exchange of components of the security keys that are used for media encryption. The more secure key management solutions involve establishment of the keys using a key negotiation technique wherein each end of the call provides one half of the component of the key (this method is commonly known as dynamic key exchange (DKE) and employed in key management protocols such as MIKEY option 3 or SDescription).
These mechanisms require high amount of processing capacity for the originating device if a call is forked (multiple recipients are called) since the originating party must negotiate the key independently with each called device. As such these mechanisms work well for one-to-one call scenarios but not for one-to-many call scenarios like parallel ringing, pickup groups, multiple line appearances, etc. In forking scenarios the call is presented to many parties and the first one to answer determines where the call media will be established. Since SIP phones have limited processing power and SIP servers (B2BUA) do not expose multiple dialogs towards the originator these mechanisms cannot be implemented.
Therefore, it is desirable to have a system that allows a caller to initiate a secure call to multiple users over a SIP network.